This invention relates in general to telephony communication networks and more particularly, it is directed to providing a system capable of hi-fidelity audio transmission over existing switched circuit and packet networks.
The public switched telephone network (PSTN) was designed during the 1970s. It operates by converting a user""s voice to digital information by sampling the analog waveform, transmitting the digital information through the network, and converting it back to an analog waveform at the destination location. A subscriber line interface card (SLIC) typically samples the subscriber""s signal at the central office (CO) or other similar facility such as a private branch exchange (PBX). The SLIC samples the audio waveform at a rate of 8 kHz, and produces eight-bit octets for each sample using coding schemes known as xcexc-law or A-law, resulting in a total bit rate of 64 KBPS. These data bits are transmitted through the network, and at the distant end location the remote SLIC reconstructs the analog signal for analog transmission to the remote subscriber. The sampling rate and quantization scheme used by SLIC devices were chosen to provide acceptable quality voice transmission.
The 8 kHz sampling rate used by the PSTN provides reconstruction of signals up to approximately 4 kHz. However, the human ear is capable of discerning signals with energy content up to approximately 20 kHz, and the human voice can generate sounds up to about 10 kHz. The PSTN audio transmission capabilities, while adequate for subscriber-to-subscriber communication and message comprehension, are well below human capacities with respect to dynamic range and frequency content.
In the existing PSTN, each telephone or other communication device (xe2x80x9csubscriber unitxe2x80x9d) is typically interconnected by a pair of wires (xe2x80x9ctipxe2x80x9d and xe2x80x9cringxe2x80x9d wires or, cooperatively, xe2x80x9csubscriber lines,xe2x80x9d xe2x80x9csubscriber loopxe2x80x9d or xe2x80x9cphone linesxe2x80x9d) through a series of equipment to a switch at a local telephone CO. At the CO, the tip and ring lines are interconnected to a SLIC, which provides required functionality to the subscriber unit. The switches at the central offices are interconnected to provide a network of switches thereby providing communications between, e.g., a local subscriber and a remote subscriber.
The functions served by the SLIC are commonly referred to as BORSCHT functionality, an acronym for Battery supply (or talking battery), Overvoltage protection, Ringing current supply, Supervision of subscriber terminal, Coder and decoder (Codec), Hybrid (2-wire to 4-wire conversion), and Testing. Generally, these functions include talking battery, ring voltage, ring trip, off-hook detection, and call progress signals such as ringback, busy, and dial tone. In many business office environments, the small-scale function of a central office is assumed by a PBX system, which, in turn, may include a number of SLICs to provide interconnected subscriber units with the required functionality.
Analog subscriber units generate and receive analog signals. While most modern telephone networks are digital and route digital signals from point to point, the subscriber units are still predominantly analog. This is possible because, as mentioned above, the analog signal generated by an analog subscriber unit is converted to digital form for transmission through the network, and is converted at the remote location back to analog form for transmission over the subscriber loop for reception by a remote analog subscriber unit. The analog signals generated and received by analog subscriber units generally take the form of voice frequency signals for end-to-end communications between local and remote subscriber units. Such signals generally represent human speech, or may be modulated signals which are treated by the phone network as if they were ordinary speech signals.
Other analog signals generated or received at the analog subscriber units are supervisory signals that are not intended for transmission to a remote terminal. Rather, they are designed to communicate with the network to enable functions such as call initiation, call progress indication, and call termination. These signals include those provided by, or through, a SLIC such as (i) xe2x80x9ctalk battery voltagexe2x80x9d which provides power to the analog subscriber unit, (ii) xe2x80x9cring voltagexe2x80x9d which is a relatively high voltage indicative of an incoming call, (iii) call progress tones such as dial tone, busy tone, and ringback tone. These various signals and tones will be described below.
The subscriber line interface circuit provides DC power, or xe2x80x9ctalk batteryxe2x80x9d power, along the phone lines to enable operation of circuitry in subscriber units connected to those lines. Most analog telephone systems work on DC (direct current) power. Typically, the talking battery voltage on analog phone lines is between 5 (off-hook) and 48 volts (on-hook). Talking battery must provide sufficient voltage to enable analog telephones to perform functions such as amplification and sound pickup as well as other phone circuits such as DTMF keypads and speakerphone operation. The talking battery power supply should always be available to an analog telephone, in the event the phone is placed in an off-hook, or closed-circuit state.
A subscriber line interface circuit provides a ring voltage signal to the analog subscriber unit to cause the analog phone to ring in the event of an incoming telephone call. Analog phone systems recognize a ring voltage signal placed on the phone lines by the SLIC, and in turn generate an audible electronic or mechanical ring sound to alert the subscriber of an incoming call. In order to ensure that an analog phone will recognize the ring voltage signal, the ring voltage is required to be 70 to 90 volts (or 140 to 180 volts peak-to-peak AC) at a frequency of 17 to 20 Hz.
A SLIC also passes call progress tones such as dial tone, busy tone, and ringback tone to the subscriber unit. For the convenience of the subscriber who is initiating the call, these tones are provided by the central office as an indication of call status. When the calling subscriber lifts the handset, or when the subscriber unit otherwise generates an xe2x80x9coff hookxe2x80x9d condition, the central office generates a dial tone and supplies it to the calling subscriber unit to indicate the availability of phone service. After the calling subscriber has dialed a phone number of the remote (answering) subscriber unit, the SLIC passes a ring back sound directed to the calling subscriber to indicate that the network is taking action to signal the remote subscriber, i.e., that the remote subscriber is being rung. Alternatively, if the network determines that the remote subscriber unit is engaged in another call (or is already off-hook), the network generates a busy tone directed to the calling subscriber unit.
The SLIC also acts to identify the status of, or interpret signals generated by, the analog subscriber unit. For example, the SLIC provides xe2x88x9248 volts on the ring line, and 0 volts on the tip line, to the subscriber unit. The analog subscriber unit provides an open circuit when in the on-hook condition. In a xe2x80x9cloop startxe2x80x9d circuit, the analog subscriber unit generates an off-hook condition by providing a termination, i.e., by closing, or xe2x80x9cloopingxe2x80x9d the tip and ring to form a complete electrical circuit. This off-hook condition is detected by the SLIC (whereupon a dial tone is provided to the subscriber). Most residential circuits are configured as loop start circuits. Some countries, however, have other requirements. Germany, for example, requires a ground to be applied on an additional lead that acts as a control signal.
The SLIC must also be able to detect the off-hook condition during application of ring voltage. That is, when a call is incoming and the SLIC is providing the high amplitude ring voltage signal to the analog subscriber unit, the SLIC must be able to detect when the analog subscriber unit goes off-hook to answer the call. This is known as xe2x80x9cring trip.xe2x80x9d The SLIC must immediately cease the ring voltage signal upon detection of the off-hook condition, and provide the analog subscriber unit with the voice channel signals originating from the distant end subscriber unit.
The SLIC must also pass Dual Tone Multi-Frequency (DTMF) signals generated by the analog subscriber unit to the network. This signaling format is a well known method of providing dialing information. Each number on a keypad array is represented by two separate tones, one tone identifying the column, and the other representing the row. Together, two tones uniquely identify a digit. These tones are passed along to the network by the SLIC.
The PSTN has also been used to convey pure data signals. This has been done successfully by using voice-grade modems such as V.34, pure digital DDS services, and hybrid services such as V.90 and V.91. Of course, because the PSTN infrastructure was designed with the digital channel capacity of 64 Kbps in mind, the PSTN is inherently limited in its data throughput rate and has become increasingly insufficient to handle today""s data communication needs. The development and widespread use of alternative data communication networks, such as those utilized by the internet, which is a world-wide network of computers over high-capacity communication links, has brought with it an expansion in the uses for such networks. Recently, the transmission of voice data over such high-speed data networks has become quite popular.
Present day IP telephones transport voice over a network using data packets instead of circuit switched connections over voice only networks. IP Telephony refers to the transfer of voice over the Internet Protocol (IP) of the TCP/IP protocol suite. The terms Voice over IP (VoIP) or xe2x80x9cIP Telephonyxe2x80x9d are generally used when referring to voice over any packet network. IP Telephones originally existed in the form of client software running on multimedia PCs and provided low-cost communications over the Internet. These systems suffered from low quality voice communication due largely to the intermittent data throughput rates, resulting in excessive delay, variable delay and network congestion resulting in lost packets.
Even in such modem data communication systems capable of high throughput connections, by design, the quality of the voice signals are only comparable to that obtained with the legacy 64 Kbps xcexc-law and A-law coding schemes. Disclosed herein is a system that provides high-fidelity voice/audio transmission that overcomes the sound quality limitations associated with the existing PSTN communication system and existing VoIP systems.
In accordance with an illustrative embodiment of the present invention, a high-fidelity voice/audio communication system is provided. According to an aspect of the illustrative embodiment, a high-fidelity SLIC (HSLIC) device is provided that combines traditional BORSCHT functionality with high fidelity sampling and compression techniques. The HSLIC preferably resides on a single plug-in line card contained within a multi-card chassis. The line card includes an analog interface that connects to a two-wire subscriber line, a high fidelity codec for sampling the analog signal at a high resolution and converting high rate digital signals to an analog signal, a voice processing client running on a microprocessor and associated digital memory. The voice-processing client forms data packets and forwards them, via a network interface circuit, to a communication network for transmission to a distant end subscriber. The high fidelity codec preferably has a sample rate of at least twenty thousand samples per second, and no less than 4096 quantization levels, and up to a sampling rate of 44.1 kHz with 65,536 quantization levels.
The voice processing client preferably includes an Internet Protocol (IP) processing client, Session Initiation Protocol (SIP) client, a Real Time Protocol (RTP) client, and other components of a communication protocol stack for establishing a connection over a packet based network by way of the network interface circuit, such as UDP and IP. The high fidelity line card preferably provides line voltage and signals to operate a standard POTS subscriber device, where the standard POTS subscriber device may or may not provide enhanced sound characteristics, such as a broadband frequency microphone and speaker. The data packets preferably contain compressed data samples, where the compression algorithm, is a negotiated configuration parameter. Further parameters include the codec type and IP address.
The line card establishes a high fidelity audio connection by obtaining the address of the called party; sending an invite request to the called party; receiving an okay signal indicating that the request was received; sending an acknowledge signal; quantizing audio information at a sampling rate greater than twenty thousand samples per second with a resolution of no less than 4096 quantization levels; and, packetizing the quantized data for transmission to a remote device.